If a voice call has missing data it makes it hard to understand the speaker. Therefore, we commonly allow time for a limited number of retransmissions before playing the sound to the listener. If the network path (in each direction) has total packetization delay of 15ms, total propagation delay of 25ms, and queuing delay varying between 0ms and 10ms, how large (in milliseconds) does the playback buffer need to be if we want to allow for one retransmission?

Video and audio packets needs to be reliable in a network, so a connection oriented protocol like TCP (transmission control protocol) is always configured for this case.

Delays in the transmission of packets can be noticeable with regards to videos and audios and they can be calculated and configured when necessary.

In this scenario, there is a re-transmission of a dropped packet, so it is not acknowledged by the receiver. An ICMP message is sent to the source with a delay of;

total delay to source: packetization delay + propagation delay + queuing delay

= 15ms + 25ms + 10ms = 50ms

Re-transmitting to the receiver, in equal condition, takes the same amount of time

Total delay after re-transmission = 2 × 50 = 100ms.

So the maximum delay is 100ms